Fedora 20 Update: asterisk-11.9.0-1.fc20
updates at fedoraproject.org
updates at fedoraproject.org
Thu May 8 10:17:53 UTC 2014
--------------------------------------------------------------------------------
Fedora Update Notification
FEDORA-2014-5738
2014-04-29 03:22:24
--------------------------------------------------------------------------------
Name : asterisk
Product : Fedora 20
Version : 11.9.0
Release : 1.fc20
URL : http://www.asterisk.org/
Summary : The Open Source PBX
Description :
Asterisk is a complete PBX in software. It runs on Linux and provides
all of the features you would expect from a PBX and more. Asterisk
does voice over IP in three protocols, and can interoperate with
almost all standards-based telephony equipment using relatively
inexpensive hardware.
--------------------------------------------------------------------------------
Update Information:
The Asterisk Development Team has announced the release of Asterisk 11.9.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 11.9.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following are the issues resolved in this release:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-22790 - check_modem_rate() may return incorrect rate
for V.27 (Reported by Paolo Compagnini)
* ASTERISK-23034 - [patch] manager Originate doesn't abort on
failed format_cap allocation (Reported by Corey Farrell)
* ASTERISK-23061 - [Patch] 'textsupport' setting not mentioned in
sip.conf.sample (Reported by Eugene)
* ASTERISK-23028 - [patch] Asterisk man pages contains unquoted
minus signs (Reported by Jeremy Lainé)
* ASTERISK-23046 - Custom CDR fields set during a GoSUB called
from app_queue are not inserted (Reported by Denis Pantsyrev)
* ASTERISK-23027 - [patch] Spelling typo "transfered" instead of
"transferred" (Reported by Jeremy Lainé)
* ASTERISK-23008 - Local channels loose CALLERID name when DAHDI
channel connects (Reported by Michael Cargile)
* ASTERISK-23100 - [patch] In chan_mgcp the ident in transmitted
request and request queue may differ - fix for locking (Reported
by adomjan)
* ASTERISK-22988 - [patch]T38 , SIP 488 after Rejecting image
media offer due to invalid or unsupported syntax (Reported by
adomjan)
* ASTERISK-22861 - [patch]Specifying a null time as parameter to
GotoIfTime or ExecIfTime causes segmentation fault (Reported by
Sebastian Murray-Roberts)
* ASTERISK-17837 - extconfig.conf - Maximum Include level (1)
exceeded (Reported by pz)
* ASTERISK-22662 - Documentation fix? - queues.conf says
persistentmembers defaults to yes, it appears to lie (Reported
by Rusty Newton)
* ASTERISK-23134 - [patch] res_rtp_asterisk port selection cannot
handle selinux port restrictions (Reported by Corey Farrell)
* ASTERISK-23220 - STACK_PEEK function with no arguments causes
crash/core dump (Reported by James Sharp)
* ASTERISK-19773 - Asterisk crash on issuing Asterisk-CLI 'reload'
command multiple times on cli_aliases (Reported by Joel Vandal)
* ASTERISK-22757 - segfault in res_clialiases.so on reload when
mapping "module reload" command (Reported by Gareth Blades)
* ASTERISK-17727 - [patch] TLS doesn't get all certificate chain
(Reported by LN)
* ASTERISK-23178 - devicestate.h: device state setting functions
are documented with the wrong return values (Reported by
Jonathan Rose)
* ASTERISK-23232 - LocalBridge AMI Event LocalOptimization value
is opposite to what's expected (Reported by Leon Roy)
* ASTERISK-23098 - [patch]possible null pointer dereference in
format.c (Reported by Marcello Ceschia)
* ASTERISK-23297 - Asterisk 12, pbx_config.so segfaults if
res_parking.so is not loaded, or if res_parking.conf has no
configuration (Reported by CJ Oster)
* ASTERISK-23069 - Custom CDR variable not recorded when set in
macro called from app_queue (Reported by Bryan Anderson)
* ASTERISK-19499 - ConfBridge MOH is not working for transferee
after attended transfer (Reported by Timo Teräs)
* ASTERISK-23261 - [patch]Output mixup in
${CHANNEL(rtpqos,audio,all)} (Reported by rsw686)
* ASTERISK-23279 - [patch]Asterisk doesn't support the dynamic
payload change in rtp mapping in the 200 OK response (Reported
by NITESH BANSAL)
* ASTERISK-23255 - UUID included for Redhat, but missing for
Debian distros in install_prereq script (Reported by Rusty
Newton)
* ASTERISK-23260 - [patch]ForkCDR v option does not keep CDR
variables for subsequent records (Reported by zvision)
* ASTERISK-23141 - Asterisk crashes on Dial(), in
pbx_find_extension at pbx.c (Reported by Maxim)
* ASTERISK-23336 - Asterisk warning "Don't know how to indicate
condition 33 on ooh323c" on outgoing calls from H323 to SIP peer
(Reported by Alexander Semych)
* ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
to minrate=2400, then res_fax refuse to load (Reported by David
Brillert)
* ASTERISK-23135 - Crash - segfault in ast_channel_hangupcause_set
- probably introduced in 11.7.0 (Reported by OK)
* ASTERISK-23323 - [patch]chan_sip: missing p->owner checks in
handle_response_invite (Reported by Walter Doekes)
* ASTERISK-23406 - [patch]Fix typo in "sip show peer" (Reported by
ibercom)
* ASTERISK-23310 - bridged channel crashes in bridge_p2p_rtp_write
(Reported by Jeremy Lainé)
* ASTERISK-22911 - [patch]Asterisk fails to resume WebRTC call
from hold (Reported by Vytis Valentinavičius)
* ASTERISK-23104 - Specifying the SetVar AMI without a Channel
cause Asterisk to crash (Reported by Joel Vandal)
* ASTERISK-21930 - [patch]WebRTC over WSS is not working.
(Reported by John)
* ASTERISK-23383 - Wrong sense test on stat return code causes
unchanged config check to break with include files. (Reported by
David Woolley)
* ASTERISK-20149 - Crash when faxing SIP to SIP with strictrtp set
to yes (Reported by Alexandr Gordeev)
* ASTERISK-17523 - Qualify for static realtime peers does not work
(Reported by Maciej Krajewski)
* ASTERISK-21406 - [patch] chan_sip deadlock on monlock between
unload_module and do_monitor (Reported by Corey Farrell)
* ASTERISK-23373 - [patch]Security: Open FD exhaustion with
chan_sip Session-Timers (Reported by Corey Farrell)
* ASTERISK-23340 - Security Vulnerability: stack allocation of
cookie headers in loop allows for unauthenticated remote denial
of service attack (Reported by Matt Jordan)
* ASTERISK-23311 - Manager - MoH Stop Event fails to show up when
leaving Conference (Reported by Benjamin Keith Ford)
* ASTERISK-23420 - [patch]Memory leak in manager_add_filter
function in manager.c (Reported by Etienne Lessard)
* ASTERISK-23488 - Logic error in callerid checksum processing
(Reported by Russ Meyerriecks)
* ASTERISK-23461 - Only first user is muted when joining
confbridge with 'startmuted=yes' (Reported by Chico Manobela)
* ASTERISK-20841 - fromdomain not honored on outbound INVITE
request (Reported by Kelly Goedert)
* ASTERISK-22079 - Segfault: INTERNAL_OBJ (user_data=0x6374652f)
at astobj2.c:120 (Reported by Jamuel Starkey)
* ASTERISK-23509 - [patch]SayNumber for Polish language tries to
play empty files for numbers divisible by 100 (Reported by
zvision)
* ASTERISK-23103 - [patch]Crash in ast_format_cmp, in ao2_find
(Reported by JoshE)
* ASTERISK-23391 - Audit dialplan function usage of channel
variable (Reported by Corey Farrell)
* ASTERISK-23548 - POST to ARI sometimes returns no body on
success (Reported by Scott Griepentrog)
* ASTERISK-23460 - ooh323 channel stuck if call is placed directly
and gatekeeper is not available (Reported by Dmitry Melekhov)
Improvements made in this release:
-----------------------------------
* ASTERISK-22980 - [patch]Allow building cdr_radius and cel_radius
against libfreeradius-client (Reported by Jeremy Lainé)
* ASTERISK-22661 - Unable to exit ChanSpy if spied channel does
not have a call in progress (Reported by Chris Hillman)
* ASTERISK-23099 - [patch] WSS: enable ast_websocket_read()
function to read the whole available data at first and then wait
for any fragmented packets (Reported by Thava Iyer)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.9.0
--------------------------------------------------------------------------------
ChangeLog:
* Wed Apr 23 2014 Jeffrey Ollie <jeff at ocjtech.us> - 11.9.0-1:
- The Asterisk Development Team has announced the release of Asterisk 11.9.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- * ASTERISK-22790 - check_modem_rate() may return incorrect rate
- for V.27 (Reported by Paolo Compagnini)
- * ASTERISK-23034 - [patch] manager Originate doesn't abort on
- failed format_cap allocation (Reported by Corey Farrell)
- * ASTERISK-23061 - [Patch] 'textsupport' setting not mentioned in
- sip.conf.sample (Reported by Eugene)
- * ASTERISK-23028 - [patch] Asterisk man pages contains unquoted
- minus signs (Reported by Jeremy Lainé)
- * ASTERISK-23046 - Custom CDR fields set during a GoSUB called
- from app_queue are not inserted (Reported by Denis Pantsyrev)
- * ASTERISK-23027 - [patch] Spelling typo "transfered" instead of
- "transferred" (Reported by Jeremy Lainé)
- * ASTERISK-23008 - Local channels loose CALLERID name when DAHDI
- channel connects (Reported by Michael Cargile)
- * ASTERISK-23100 - [patch] In chan_mgcp the ident in transmitted
- request and request queue may differ - fix for locking (Reported
- by adomjan)
- * ASTERISK-22988 - [patch]T38 , SIP 488 after Rejecting image
- media offer due to invalid or unsupported syntax (Reported by
- adomjan)
- * ASTERISK-22861 - [patch]Specifying a null time as parameter to
- GotoIfTime or ExecIfTime causes segmentation fault (Reported by
- Sebastian Murray-Roberts)
- * ASTERISK-17837 - extconfig.conf - Maximum Include level (1)
- exceeded (Reported by pz)
- * ASTERISK-22662 - Documentation fix? - queues.conf says
- persistentmembers defaults to yes, it appears to lie (Reported
- by Rusty Newton)
- * ASTERISK-23134 - [patch] res_rtp_asterisk port selection cannot
- handle selinux port restrictions (Reported by Corey Farrell)
- * ASTERISK-23220 - STACK_PEEK function with no arguments causes
- crash/core dump (Reported by James Sharp)
- * ASTERISK-19773 - Asterisk crash on issuing Asterisk-CLI 'reload'
- command multiple times on cli_aliases (Reported by Joel Vandal)
- * ASTERISK-22757 - segfault in res_clialiases.so on reload when
- mapping "module reload" command (Reported by Gareth Blades)
- * ASTERISK-17727 - [patch] TLS doesn't get all certificate chain
- (Reported by LN)
- * ASTERISK-23178 - devicestate.h: device state setting functions
- are documented with the wrong return values (Reported by
- Jonathan Rose)
- * ASTERISK-23232 - LocalBridge AMI Event LocalOptimization value
- is opposite to what's expected (Reported by Leon Roy)
- * ASTERISK-23098 - [patch]possible null pointer dereference in
- format.c (Reported by Marcello Ceschia)
- * ASTERISK-23297 - Asterisk 12, pbx_config.so segfaults if
- res_parking.so is not loaded, or if res_parking.conf has no
- configuration (Reported by CJ Oster)
- * ASTERISK-23069 - Custom CDR variable not recorded when set in
- macro called from app_queue (Reported by Bryan Anderson)
- * ASTERISK-19499 - ConfBridge MOH is not working for transferee
- after attended transfer (Reported by Timo Teräs)
- * ASTERISK-23261 - [patch]Output mixup in
- ${CHANNEL(rtpqos,audio,all)} (Reported by rsw686)
- * ASTERISK-23279 - [patch]Asterisk doesn't support the dynamic
- payload change in rtp mapping in the 200 OK response (Reported
- by NITESH BANSAL)
- * ASTERISK-23255 - UUID included for Redhat, but missing for
- Debian distros in install_prereq script (Reported by Rusty
- Newton)
- * ASTERISK-23260 - [patch]ForkCDR v option does not keep CDR
- variables for subsequent records (Reported by zvision)
- * ASTERISK-23141 - Asterisk crashes on Dial(), in
- pbx_find_extension at pbx.c (Reported by Maxim)
- * ASTERISK-23336 - Asterisk warning "Don't know how to indicate
- condition 33 on ooh323c" on outgoing calls from H323 to SIP peer
- (Reported by Alexander Semych)
- * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
- to minrate=2400, then res_fax refuse to load (Reported by David
- Brillert)
- * ASTERISK-23135 - Crash - segfault in ast_channel_hangupcause_set
- - probably introduced in 11.7.0 (Reported by OK)
- * ASTERISK-23323 - [patch]chan_sip: missing p->owner checks in
- handle_response_invite (Reported by Walter Doekes)
- * ASTERISK-23406 - [patch]Fix typo in "sip show peer" (Reported by
- ibercom)
- * ASTERISK-23310 - bridged channel crashes in bridge_p2p_rtp_write
- (Reported by Jeremy Lainé)
- * ASTERISK-22911 - [patch]Asterisk fails to resume WebRTC call
- from hold (Reported by Vytis Valentinavičius)
- * ASTERISK-23104 - Specifying the SetVar AMI without a Channel
- cause Asterisk to crash (Reported by Joel Vandal)
- * ASTERISK-21930 - [patch]WebRTC over WSS is not working.
- (Reported by John)
- * ASTERISK-23383 - Wrong sense test on stat return code causes
- unchanged config check to break with include files. (Reported by
- David Woolley)
- * ASTERISK-20149 - Crash when faxing SIP to SIP with strictrtp set
- to yes (Reported by Alexandr Gordeev)
- * ASTERISK-17523 - Qualify for static realtime peers does not work
- (Reported by Maciej Krajewski)
- * ASTERISK-21406 - [patch] chan_sip deadlock on monlock between
- unload_module and do_monitor (Reported by Corey Farrell)
- * ASTERISK-23373 - [patch]Security: Open FD exhaustion with
- chan_sip Session-Timers (Reported by Corey Farrell)
- * ASTERISK-23340 - Security Vulnerability: stack allocation of
- cookie headers in loop allows for unauthenticated remote denial
- of service attack (Reported by Matt Jordan)
- * ASTERISK-23311 - Manager - MoH Stop Event fails to show up when
- leaving Conference (Reported by Benjamin Keith Ford)
- * ASTERISK-23420 - [patch]Memory leak in manager_add_filter
- function in manager.c (Reported by Etienne Lessard)
- * ASTERISK-23488 - Logic error in callerid checksum processing
- (Reported by Russ Meyerriecks)
- * ASTERISK-23461 - Only first user is muted when joining
- confbridge with 'startmuted=yes' (Reported by Chico Manobela)
- * ASTERISK-20841 - fromdomain not honored on outbound INVITE
- request (Reported by Kelly Goedert)
- * ASTERISK-22079 - Segfault: INTERNAL_OBJ (user_data=0x6374652f)
- at astobj2.c:120 (Reported by Jamuel Starkey)
- * ASTERISK-23509 - [patch]SayNumber for Polish language tries to
- play empty files for numbers divisible by 100 (Reported by
- zvision)
- * ASTERISK-23103 - [patch]Crash in ast_format_cmp, in ao2_find
- (Reported by JoshE)
- * ASTERISK-23391 - Audit dialplan function usage of channel
- variable (Reported by Corey Farrell)
- * ASTERISK-23548 - POST to ARI sometimes returns no body on
- success (Reported by Scott Griepentrog)
- * ASTERISK-23460 - ooh323 channel stuck if call is placed directly
- and gatekeeper is not available (Reported by Dmitry Melekhov)
-
- Improvements made in this release:
- -----------------------------------
- * ASTERISK-22980 - [patch]Allow building cdr_radius and cel_radius
- against libfreeradius-client (Reported by Jeremy Lainé)
- * ASTERISK-22661 - Unable to exit ChanSpy if spied channel does
- not have a call in progress (Reported by Chris Hillman)
- * ASTERISK-23099 - [patch] WSS: enable ast_websocket_read()
- function to read the whole available data at first and then wait
- for any fragmented packets (Reported by Thava Iyer)
* Tue Mar 11 2014 Jeffrey Ollie <jeff at ocjtech.us> - 11.8.1-1:
- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.15, 11.6, and Asterisk 1.8, 11, and 12. The available security
- releases are released as versions 1.8.15-cert5, 11.6-cert2, 1.8.26.1, 11.8.1,
- and 12.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolve the following issues:
-
- * AST-2014-001: Stack overflow in HTTP processing of Cookie headers.
-
- Sending a HTTP request that is handled by Asterisk with a large number of
- Cookie headers could overflow the stack.
-
- Another vulnerability along similar lines is any HTTP request with a
- ridiculous number of headers in the request could exhaust system memory.
-
- * AST-2014-002: chan_sip: Exit early on bad session timers request
-
- This change allows chan_sip to avoid creation of the channel and
- consumption of associated file descriptors altogether if the inbound
- request is going to be rejected anyway.
-
- Additionally, the release of 12.1.1 resolves the following issue:
-
- * AST-2014-003: res_pjsip: When handling 401/407 responses don't assume a
- request will have an endpoint.
-
- This change removes the assumption that an outgoing request will always
- have an endpoint and makes the authenticate_qualify option work once again.
-
- Finally, a security advisory, AST-2014-004, was released for a vulnerability
- fixed in Asterisk 12.1.0. Users of Asterisk 12.0.0 are encouraged to upgrade to
- 12.1.1 to resolve both vulnerabilities.
-
- These issues and their resolutions are described in the security advisories.
-
- For more information about the details of these vulnerabilities, please read
- security advisories AST-2014-001, AST-2014-002, AST-2014-003, and AST-2014-004,
- which were released at the same time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.15-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.26.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.8.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.1.1
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2014-001.pdf
- * http://downloads.asterisk.org/pub/security/AST-2014-002.pdf
- * http://downloads.asterisk.org/pub/security/AST-2014-003.pdf
- * http://downloads.asterisk.org/pub/security/AST-2014-004.pdf
* Tue Mar 4 2014 Jeffrey Ollie <jeff at ocjtech.us> - 11.8.0-1:
- The Asterisk Development Team has announced the release of Asterisk 11.8.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- * ASTERISK-22544 - Italian prompt vm-options has advertisement in
- it (Reported by Rusty Newton)
- * ASTERISK-21383 - STUN Binding Requests Not Being Sent Back from
- Asterisk to Chrome (Reported by Shaun Clark)
- * ASTERISK-22478 - [patch]Can't use pound(hash) symbol for custom
- DTMF menus in ConfBridge (processed as directive) (Reported by
- Nicolas Tanski)
- * ASTERISK-12117 - chan_sip creates a new local tag (from-tag) for
- every register message (Reported by Pawel Pierscionek)
- * ASTERISK-20862 - Asterisk min and max member penalties not
- honored when set with 0 (Reported by Schmooze Com)
- * ASTERISK-22746 - [patch]Crash in chan_dahdi during caller id
- read (Reported by Michael Walton)
- * ASTERISK-22788 - [patch] main/translate.c: access to variable f
- after free in ast_translate() (Reported by Corey Farrell)
- * ASTERISK-21242 - Segfault when T.38 re-invite retransmission
- receives 200 OK (Reported by Ashley Winters)
- * ASTERISK-22590 - BufferOverflow in unpacksms16() when receiving
- 16 bit multipart SMS with app_sms (Reported by Jan Juergens)
- * ASTERISK-22905 - Prevent Asterisk functions that are 'dangerous'
- from being executed from external interfaces (Reported by Matt
- Jordan)
- * ASTERISK-23021 - Typos in code : "avaliable" instead of
- "available" (Reported by Jeremy Lainé)
- * ASTERISK-22970 - [patch]Documentation fix for QUOTE() (Reported
- by Gareth Palmer)
- * ASTERISK-21960 - ooh323 channels stuck (Reported by Dmitry
- Melekhov)
- * ASTERISK-22350 - DUNDI - core dump on shutdown - segfault in
- sqlite3_reset from /usr/lib/libsqlite3.so.0 (Reported by Birger
- "WIMPy" Harzenetter)
- * ASTERISK-22942 - [patch] - Asterisk crashed after
- Set(FAXOPT(faxdetect)=t38) (Reported by adomjan)
- * ASTERISK-22856 - [patch]SayUnixTime in polish reads minutes
- instead of seconds (Reported by Robert Mordec)
- * ASTERISK-22854 - [patch] - Deadlock between cel_pgsql unload and
- core_event_dispatcher taskprocessor thread (Reported by Etienne
- Lessard)
- * ASTERISK-22910 - [patch] - REPLACE() calls strcpy on overlapping
- memory when <replace-char> is empty (Reported by Gareth Palmer)
- * ASTERISK-22871 - cel_pgsql module not loading after "reload" or
- "reload cel_pgsql.so" command (Reported by Matteo)
- * ASTERISK-23084 - [patch]rasterisk needlessly prints the
- AST-2013-007 warning (Reported by Tzafrir Cohen)
- * ASTERISK-17138 - [patch] Asterisk not re-registering after it
- receives "Forbidden - wrong password on authentication"
- (Reported by Rudi)
- * ASTERISK-23011 - [patch]configure.ac and pbx_lua don't support
- lua 5.2 (Reported by George Joseph)
- * ASTERISK-22834 - Parking by blind transfer when lot full orphans
- channels (Reported by rsw686)
- * ASTERISK-23047 - Orphaned (stuck) channel occurs during a failed
- SIP transfer to parking space (Reported by Tommy Thompson)
- * ASTERISK-22946 - Local From tag regression with sipgate.de
- (Reported by Stephan Eisvogel)
- * ASTERISK-23010 - No BYE message sent when sip INVITE is received
- (Reported by Ryan Tilton)
- * ASTERISK-23135 - Crash - segfault in ast_channel_hangupcause_set
- - probably introduced in 11.7.0 (Reported by OK)
-
- Improvements made in this release:
- -----------------------------------
- * ASTERISK-22728 - [patch] Improve Understanding Of 'Forcerport'
- When Running "sip show peers" (Reported by Michael L. Young)
- * ASTERISK-22659 - Make a new core and extra sounds release
- (Reported by Rusty Newton)
- * ASTERISK-22919 - core show channeltypes slicing (Reported by
- outtolunc)
- * ASTERISK-22918 - dahdi show channels slices PRI channel dnid on
- output (Reported by outtolunc)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.8.0
* Sat Dec 28 2013 Jeffrey Ollie <jeff at ocjtech.us> - 11.7.0-1:
- The Asterisk Development Team has announced the release of Asterisk 11.7.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is a sample of the issues resolved in this release:
-
- * --- app_confbridge: Can now set the language used for announcements
- to the conference.
- (Closes issue ASTERISK-19983. Reported by Jonathan White)
-
- * --- app_queue: Fix CLI "queue remove member" queue_log entry.
- (Closes issue ASTERISK-21826. Reported by Oscar Esteve)
-
- * --- chan_sip: Do not increment the SDP version between 183 and 200
- responses.
- (Closes issue ASTERISK-21204. Reported by NITESH BANSAL)
-
- * --- chan_sip: Allow a sip peer to accept both AVP and AVPF calls
- (Closes issue ASTERISK-22005. Reported by Torrey Searle)
-
- * --- chan_sip: Fix Realtime Peer Update Problem When Un-registering
- And Expires Header In 200ok
- (Closes issue ASTERISK-22428. Reported by Ben Smithurst)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.7.0
* Sat Dec 28 2013 Jeffrey Ollie <jeff at ocjtech.us> - 11.6.1-1:
- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.15, 11.2, and Asterisk 1.8, 10, and 11. The available security
- releases are released as versions 1.8.15-cert4, 11.2-cert3, 1.8.24.1, 10.12.4,
- 10.12.4-digiumphones, and 11.6.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolve the following issues:
-
- * A buffer overflow when receiving odd length 16 bit messages in app_sms. An
- infinite loop could occur which would overwrite memory when a message is
- received into the unpacksms16() function and the length of the message is an
- odd number of bytes.
-
- * Prevent permissions escalation in the Asterisk Manager Interface. Asterisk
- now marks certain individual dialplan functions as 'dangerous', which will
- inhibit their execution from external sources.
-
- A 'dangerous' function is one which results in a privilege escalation. For
- example, if one were to read the channel variable SHELL(rm -rf /) Bad
- Things(TM) could happen; even if the external source has only read
- permissions.
-
- Execution from external sources may be enabled by setting 'live_dangerously'
- to 'yes' in the [options] section of asterisk.conf. Although doing so is not
- recommended.
-
- These issues and their resolutions are described in the security advisories.
-
- For more information about the details of these vulnerabilities, please read
- security advisories AST-2013-006 and AST-2013-007, which were
- released at the same time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.15-cert4
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.2-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.24.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.12.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.12.4-digiumphones
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.6.1
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2013-006.pdf
- * http://downloads.asterisk.org/pub/security/AST-2013-007.pdf
* Sat Dec 28 2013 Jeffrey Ollie <jeff at ocjtech.us> - 11.6.0-1:
- The Asterisk Development Team has announced the release of Asterisk 11.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is a sample of the issues resolved in this release:
-
- * --- Confbridge: empty conference not being torn down
- (Closes issue ASTERISK-21859. Reported by Chris Gentle)
-
- * --- Let Queue wrap up time influence member availability
- (Closes issue ASTERISK-22189. Reported by Tony Lewis)
-
- * --- Fix a longstanding issue with MFC-R2 configuration that
- prevented users
- (Closes issue ASTERISK-21117. Reported by Rafael Angulo)
-
- * --- chan_iax2: Fix saving the wrong expiry time in astdb.
- (Closes issue ASTERISK-22504. Reported by Stefan Wachtler)
-
- * --- Fix segfault for certain invalid WebSocket input.
- (Closes issue ASTERISK-21825. Reported by Alfred Farrugia)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.6.0
* Mon Oct 21 2013 Jeffrey Ollie <jeff at ocjtech.us> - 11.5.1-3:
- Disable hardened build, as it's apparently causing problems loading modules.
--------------------------------------------------------------------------------
This update can be installed with the "yum" update program. Use
su -c 'yum update asterisk' at the command line.
For more information, refer to "Managing Software with yum",
available at http://docs.fedoraproject.org/yum/.
All packages are signed with the Fedora Project GPG key. More details on the
GPG keys used by the Fedora Project can be found at
https://fedoraproject.org/keys
--------------------------------------------------------------------------------
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